5.1.2 Gateway Configuration

System Parameters

General

  • Saving call data (CDR) – define the call types for which the GSM gateway shall store information in the CDR.
  • Gateway ID – provides 2N® BRI gateway with a numerical code in the CDR in case multiple devices generate CDRs in the network.
  • Number for remote control.

System Restart

  • Enable system restart  enable/disable the feature.
  • Time of system restart [hh:mm] – set the time for system restart.

 

Note

  • If there are active calls, restart will be executed 10 seconds after the end of the last call.

 

Mobility Extension (DTMF settings)

  • Start dialling (quick call forwarding) – set the DTMF code for quick call forwarding start.
  • End dialling (quick call forwarding) – set the DTMF code for quick call forwarding end.
  • Hold call – set the DTMF code for the current call holding.
  • Hang up call – set the DTMF code for the current call termination.
  • 'Follow me' activation – activate the Follow me function to make the GSM gateway start routing calls to the defined GSM/UMTS user number. The default value is *55.
  • 'Follow me' deactivation – deactivate the Follow me function. The default value is #55.
  • 'SMS at no answer' activation – activate the SMS at no answer function for a registered user. The default value is *33.
  • 'SMS at no answer' deactivation – deactivate the SMS at no answer function for a registered user. The default value is #33. 

Tip

  • Activate/deactivate the SMS at no answer and Follow me functions by dialling the above mentioned DTMF access codes into the GSM gateway from a registered mobile user number. You can change the function values via the configuration interface too (see below).

Others

  • PIN – set the PIN code for the PIN–secured SIM cards.

Caution

  • A PIN–active SIM card with a PIN value other than that set in the GSM gateway configuration will be blocked by the gateway with the 'pin–err' cause. Enter the correct PIN on your mobile phone to unlock the SIM card!
  • End of dialling (empty=off) – set the DTMF code for DTMF dialling end for DISA incoming calls. The default value is „#".

List of emergency numbers

The window displays a list of emergency numbers, which are normally routed to the BRI interface. If the BRI line is disconnected, the emergency numbers are dialled automatically via any GSM/UMTS module according to the following rules:

  • Search of a logged-in GSM/UMTS module (regardless of free minutes);
  • Search of a blocked or network searching GSM/UMTS module.

The table includes an exact format of the number to be called (112,911, etc.). The 'x' placeholder stands for any digit in the number to be called. The '{}'_ placeholder means the rest of the number. For example:

Format

Allowed numbers

123

123 only

14x0

1400,1410,1420,…1490

999_

All numbers starting with 999

LED indication

  • GSM signal mode – set the GSM module signal LED indication.
    • None
    • Module1 only
    • Module2 only
    • All modules

VoIP Parameters

VoIP functions

  • Day of deleting statistics on VoIP (every month) – set the day for automatic deletion of call statistics via the VoIP interface. None = statistics will not be deleted automatically.

SIP protocol settings

  • Use CLIP from INVITE field – define that CLIP from the 'Contact' or 'From' field shall be used for call routing to GSM/UMTS.
  • Send 180 ringing instead of 183 session progress.
  • Send 200 OK instead of 180/183.
  • Send 200 OK and BYE when rejected from GSM.
  • Send 200 OK on REGISTER request – virtual registration of device in 2N® BRI gateway (for registration requiring equipment).
  • Replace CLIP from GSM with Caller ID.
  • Deny DTMF according to RFC2833.
  • Forward DTMF for ME.

SIP registration

  • Registration expires [s] – set the expiration time for the 2N® BRI gateway registration data with SIP proxy.
  • Reattempt registration [s] – set the time interval after which the request shall be resent.
  • Registration domain (realm).
  • Caller ID.
  • Username – registration data with SIP proxy.
  • Password – registration data with SIP proxy.

Voice parameters

  • First RTP port (even: 1024 – 65524) – set the number of the first RTP port. The RTP port number must be even according to the recommendation.
  • Last RTP port (even: first RTP+10 – 65534) – set the number of the last RTP port. The RTP port number must be even according to the recommendation. The recommended minimum RTP port range is 10.

Codecs settings

Set details for the G.711a/u or G.729 codecs.

 

Codecs priority

Set the types of the speech codecs to be preferred.

  • Priority 1
  • Priority 2
  • Priority 3


IP addresses

  • SIP proxy (IP / GSM)  the SIP proxy IP address from which 2N® BRI gateway awaits the GSM outgoing call requests.

Tip

  • If you keep the default values (0.0.0.0), 2N® BRI gateway will receive requests from any IP address.
  • SIP proxy (GSM / IP) – the SIP proxy IP address to which 2N® BRI gateway turns in the case of GSM incoming calls.
  • SIP registrar – SIP registration server IP address. 

Tip

  • You can use the domain name Registration domain (realm) for the SIP proxy (IP / GSM), SIP proxy (GSM / IP) and SIP registrar IP addresses on condition that you complete the domain name Registration domain (realm) and set the DNS server address properly in the Web configuration / Ethernet configuration section. The SIP proxy and SIP registrar IP addresses must be set to the default value (0.0.0.0).
  • NAT firewall – NAT firewall IP address.
  • STUN server – STUN server IP address (Simple Traversal of UDP through NATs (Network Address Translation)) for obtaining the public IP address with which 2N® BRI gateway operates in the Internet. You are recommended to complete this field if 2N® BRI gateway is installed in a private network separated from the Internet via NAT or firewall. The pre–set port for sending requests to STUN is 3478.
  • Next STUN request (60–6553, 0=off) [s] – update of information on the 2N® BRI gateway public IP address. Use this parameter to configure the frequency of queries routed to the STUN server.

Note

  • In case the GSM gateway is installed behind the NAT, make proper routing settings in the NAT router for the relevant ports (SIP, RTP, STUN). Integrated firewalls can also affect VoIP calls!

Tip

  • Should you have call troubles (such as unilateral audibility, connection errors), make sure that all the active elements on the VoIP call route have been set properly. For easy troubleshooting, try the point–to–point connection with the software IP phone (SJ phone, e.g.) in your PC and, at the same time, apply network analyzer tracing (WireShark – www.wireshark.org).
  • Refer to Subs 5.1.7 for easy tracing by the BRI gateway.

Tones generated to VoIP

  • Ring tone to VoIP – enable generation of a user ring tone or transfer of the real ring tone from the GSM/UMTS networks.

ISDN Parameters

Use this window to set the BRI ISDN port parameters. The appearance and count of the parameters may be different in 2N® BRI Lite and 2N® BRI Enterprise due to different counts of ISDN BRI ports.

 

BRI mode selection

  • Mode – set the BRI1 and BRI2 (for 2N® BRI Enterprise only) ports.

BRI1 and BRI2

  • TEI Address – set a fixed TEI address for connection of port(s) in the Point–to–Point mode.
  • MTP – activate assignment of the dynamic TEI address (Point–to–Multipoint mode).

Progress indicator value – set the value for each progress element for call setup. Please respect the PBX and PSTN settings to avoid wrong evaluation of messages sent by the BRI gateway and, subsequently, call setup errors. Refer to the table for the decimal numbers to be assigned to the progress messages.

Number

Meaning

OFF

Progress element is not sent in the message

1

Connection is not end–to–end ISDN, following progress messages will be sent in the speech band

2

Call destination address is not ISDN

3

Call initiator address is not ISDN

4

Call is returning to ISDN

8

Communication between interconnected systems has lead to a change of the telecommunication service (for end–to–end ISDN connection only)

10

Delay due to speech interface

BRI functions

  • Day of deleting statistics on BRI (every month) – set this item to '0' to disable periodical (monthly) deleting of statistics. Set this value to 'x' other than '0' to enable deletion of statistic data on x–th day of a month.
  • Digits count in SETUP (en–block) – set the count of outgoing dialling digits to be sent by the gateway in the SETUP message in the ENBLOCK format. The remaining digits will be sent in the OVERLAP format, i.e. in the information element following the SETUP message. The OVERLAP mode is used in analogue networks.

Example:

SETUP digit count: 7, user number: 601234567

Call setup messages:

SETUP (601234567)

INFO (6)

INFO (7)

  • Receive dial number from Subaddress – use this parameter to receive dialling from the subaddress element instead of standard CDN.
  • Don't send Connect ACK on TE – use this parameter to enable/disable sending of the CONNECT ACK message to the TE port.
  • Use CLIR if requested from ISDN (SETUP) – enable automatic CLIR resending to GSM/UMTS if required so by the ISDN.

Tone signalling for calls from ISDN

  • Dial tone to BRI1 with empty SETUP – set the dial tone type to be generated by the BRI gateway.
  • Ring tone – set the ring tone type to be generated by the BRI gateway.
  • Generate busy tone to BRI1 – set the busy tone type to be generated by the BRI gateway into the BRI 1 interface.
  • Generate busy tone to BRI2 – set the busy tone type to be generated by the BRI gateway into the BRI 2 interface.

Numbering plan settings

CDN, CGN – use these parameters to set the Numbering plan for the called (CDN) and calling (CGN) numbers.

Binary value

Decimal value

Description

0000

0

Unknown numbering plan

0001

1

ISDN/Telephony numbering plan

0011

3

Sata numbering plan

0100

4

Telex numbering plan

1000

8

National standard numbering plan

1001

9

Private numbering plan

1111

15

Reserved for Extension

ISDN Parameters – Monitoring

Upozornění

  • This service is subject to licence! Refer to Management/Licence key for the current GSM gateway licence state details.

BRI1 alerts

  • Still activated ISDN layer 1 – enable/disable keeping of ISDN layer 1 active.
  • Still activated ISDN layer 2 (SABME/UA) – enable/disable keeping of ISDN layer 2 active.
  • Send SMS at state changes       enable/disable alert sending upon BRI1 state change. 


BRI2 alerts

  • Still activated ISDN layer 1 – enable/disable keeping of ISDN layer 1 active.
  • Still activated ISDN layer 2 (SABME/UA) – enable/disable keeping of ISDN layer 2 active.
  • Send SMS at state changes  –   enable/disable alert sending upon BRI1 state change.


BRI common settings

  • Timeout for ISDN line deactivation detection [s]  an SMS alert on BRI1/BRI2 deactivation is sent after this timemout.
  • Timeout for ISDN line activation detection [s] –  an SMS alert on BRI1/BRI2 activation is sent after this timeout.
  • Numbers where SMS will be sent to – list of numbers to which the SMS alert shall be sent.
  • Text of SMS  – SMS alert text with the following parameters:
    • %P(x|y), where "x" represents any text concerning BRI1 and "y" represents any text concerning BRI2. 
    • %A(x|y), where "x" represents any text for interface deactivation and "y" represents any text for interface activation.

Settings of Alive SMS interval

  • Send Alive SMS   – enable/disable sending of ALIVE SMS.
  • Time [hh:mm]  – ALIVE SMS sending time.
  • Days interval  – ALIVE SMS sending interval.
  • Numbers where SMS will be sent to   – list of numbers to which the ALIVE SMS shall be sent.
  • Text of SMS  – ALIVE SMS text.

GSM Basic Parameters

GSM selection

  • Assignment of GSM-channel
    • Cyclical
    • Locked - peer to ISDN channel
    • Smart - least used minutes
    • Smart - most remaining minutes
    • Linear - always first free module 

Caution

  • The Locked - peer to ISDN channel selection has a higher priority than the LCR table. All outgoing/incoming calls are locked in the GSM-ISDN channel pair. The GSM channel is selected cyclically for outgoing VoIP calls.
  • The Smart – least used minutes and Smart – most remaining minutes rules are useful where different limit settings are used for SMS.

Number of digits dialled from VoIP

  • Minimum numbers from VoIP – set the minimum count of digits to be dialled into the GSM network.
  • Maximum numbers from VoIP – set the maximum count of digits to be dialled into the GSM network.
  • Wait for next digit [s] – set the timeout for 2N® BRI gateway to wait for the next digit dialled from VoIP to GSM.

Calls

  • Relax timeout [s] – set the timeout between the end of the last call and beginning of the next call made via one and the same GSM module (incoming and outgoing calls are rejected during this timeout). The recommended relax value is 2 seconds – please do not change this setting unless absolutely necessary.
  • Timeout for ringing to GSM [s] – ringing timeout for outgoing calls to GSM. If not answered or terminated within this timeout, the call will be terminated automatically by the gateway when this timeout elapses.
  • Source interface for CallBack – select the CallBack source interface for CallBack routing. The LCR table must include the appropriate outgoing traffic rule: if, for example, a VoIP port is the CallBack source interface, routing from VoIP to GSM must be defined in the LCR.
  • Delay for fast CallBack [s] – set the delay between the CallBack request and outgoing call if Auto end to CallBack request is active.

Holiday list

List of days on which calls will be routed like on weekends in the LCR.

 

DTMF settings

Minimum delay between two identical DTMF characters [s/100] received.

 

Tone detector settings

The GSM gateway can automatically detect user defined tones transmitted by GSM/UMTS for setting up outgoing calls to the GSM/UMTS networks. In general, they are tones of a transferred number. For the purpose of such detection, the GSM gateway automatically terminates the call and tries to set it up via another available outgoing group (if defined in the LCR).

  • Frequency 1;2;3;4 – define the frequencies for the tone to be detected.
  • Sequence list – set the sequence of the tones to be detected.

Voice message detector settings

  • Minimum percent to match – set the match percentage range in which the voice message is detected as identical with one of the voice messages recorded in the  Gateway control / Voice messages section under index 30 - 37. The recommended value is 70-90%.

Cinterion Module Settings

  • Transmission volume [dB] – set the module volume for the outgoing direction.

  • Reception volume [dB] – set the module volume for the incoming direction

  • Enable connection tone – enable the connection tone.

  • End call with SHUP – end calls with the SHUP command.

  • Enable HR codec – enable the Half Rate codec for the GSM network.

  • Enable AMR codec – enable the Adaptive Multi-Rate codec for the GSM network.

Wavecom Module Settings

  • Transmission volume [dB] – set the module volume for the outgoing direction.

  • Reception volume [dB] – set the module volume for the incoming direction

  • GSM-band selection – select the GSM frequencies for the mobile network.
  • Echo cancelling – enable/disable echo cancelling for the module.

  • Enable HR codec – enable the Half Rate codec for the GSM network.

  • Enable AMR codec – enable the Adaptive Multi-Rate codec for the GSM network.

Telit Module Settings

  • Transmission volume [dB] – set the module volume for the outgoing direction.

  • Reception volume [dB] – set the module volume for the incoming direction

  • Type of used networks – select the network types for the module to log in.
  • Automatic band selection – enable the automatic mobile frequency band selection.
  • GSM-band selection – select the GSM frequencies for the mobile network.
  • UMTS-band selection – select the GUMTS frequencies for the mobile network.
  • Enable HR codec – enable the Half Rate codec for the GSM network.

  • Enable AMR codec – enable the Adaptive Multi-Rate codec for the GSM network.

  • Enable FR AMR Wideband codec – enable the Full Rate Adaptive Multi-Rate codec for UMTS.
  • Enable UMTS AMR Version 2 codec – enable the Full Rate Adaptive Multi-Rate version 2 codec for UMTS.
  • Enable UMTS AMR Wideband codec – enable the Adaptive Multi-Rate WideBand for UMTS.
  • Enable echo canceller – enable/disable echo cancelling for the module.
  • Enable noise reduction – enable/disable noise reduction for the module.

Warning

  • The GSM/UMTS codec and noise reduction settings of the above listed module types may affect the DTMF detection quality. Make sure that the settings comply with the mobile network used.

Audio level DSP

Set the voice level for calls in the gateway signal processor.

  • Output audio level DSP [dB] – audio volume gain/loss to VoIP
  • Input audio level DSP [dB] –  audio volume gain/loss to GSM

Caution

  • An extremely high voice level may lead to poor voice quality (distortion, echo, etc.) and wrong DTMF detection!

Tone generated for incoming calls from GSM/UMTS

  • Dial tone – set the dial tone for incoming calls from GSM/UMTS.
  • Ring tone – set the ring tone for incoming calls from GSM/UMTS.
  • Generate busy tone to GSM/UMTS – enable busy tone generation for call termination.

Caution

  • If the Generate busy tone function is enabled, the duration of outgoing calls billed by the GSM/UMTS provider will be extended!

Error GSM/UMTS causes

  • Set the ISDN release cause for each of the below mentioned statuses. Every call that meets any of the below mentioned requirements, will be rejected with a user defined cause (the ISDN cause number will be translated to VoIP as a SIP code as defined below).
    • Lack of digits in OVERLAP mode – any call that fails to meet the minimum digits count requirement will be rejected.
    • Restricted number prefix – any call whose prefix is not included in any of the prefix lists will be rejected.
    • Selected module / GSM group is not ready – a call will be rejected in case there is no available GSM module in the selected (by LCR) outgoing GSM group.
    • Selected module / GSM groups are not ready – a call will be rejected in case there is no available GSM module in the selected (by LCR) outgoing GSM groups.

Cause translation

The release cause received from GSM/UMTS can be converted into another ISDN release cause. The resultant ISDN cause number will be converted into a SIP code according to the table below:

Conversion table:

ISDN cause valueDescriptionSIP codeDescription
1Unallocated number410Gone
3No route to destination404Not found
6Channel unacceptable503Service unavailable
16Normal call clearingBYE 
17User busy486Busy here
18No user responding480Temporarily unavail.
19No answer from user480Temporarily unavail.
21Call rejected603Decline
22Number changed410Gone
27Destination out of order404Not found
28Address incomplete484Address incomplete
29Facility rejected501Not implemented
31Normal, unspecifiedBYE 
34No circuit available503Service unavailable
38Network out of order503Service unavailable
41Temporary failure503Service unavailable
42Switching equipment congestion503Service unavailable
44Requested facility not subscribed503Service unavailable
47Resource unavailable503Service unavailable
50Requested facility not subscribed503Service unavailable
55Incoming class barred within CVG603Decline
57Bearer capability not authorised501Not implemented
58Bearer cap, unavailable at present503Not implemented
63Service or option unavailable501Service unavailable
65Bearer cap, not implemented501Not implemented
79Service or option not implemented501Not implemented
87User not member of CVG603Decline
88Incompatible destination400Bad request
98Invalid message400Bad request
102Recover on timer expiry408Request timeout
XXXThe other received CAU from netw.500Internal server error

Others

  • Text of SMS at no answer – edit the text of the SMS to be sent to the calling party in case of no answer (if the function is active). The %N string will insert the CLIP received from VoIP/BRI in the SMS text.
  • Text of SMS for all calls – complete this parameter to make the GSM gateway send SMS to every called party regardless of whether or not the call was successfully connected. The %N string will insert the CLIP received from VoIP in the SMS text.
  • Save received SMS to – select the storage for received SMS messages.
  • SIM card identification – set the SIM Id (IMSI/SCID) to be entered in the CDR.
  • Disable CLIP from GSM/UMTS to VoIP – enable/disable resending of the CLIP from GSM to VoIP.
  • Reject call with CHLD – enable rejection of incoming calls from GSM/UMTS via AT+CHLD (subscriber busy) instead of standard ATH.
  • Network registration timeout [min] – set the timeout after which the module must log in. The login process will be restarted after this timeout.

GSM Group Assignment

Here you can assign the GSM/UMTS modules to groups. You can assign incoming and outgoing calls separately using the parameters below.

GSM Outgoing Groups

2N® BRI  gateway allows you to work with two groups of outgoing calls. You can set different values (call setup, called minutes and sent SMS per period, etc.) for each of them. 

 

General settings

  • Delay for CONNECT [s] – set the delay between receiving a GSM call and sending the CONNECT message.
  • Minimum ring duration to send SMS at no answer [s] – set the minimum ringing timeout for sending the SMS at no answer.

Note

  • The INVITE message must contain the called and calling numbers in order that the SMS at no answer function may work properly.
  • Delay for ALERTING [a] – set the delay before sending the ALERTING message.
    • off – no Alerting message will be forwarded.
    • real – the Alerting message will be forwarded as soon as the gateway receives Alerting from the wireless network or ring tone detectors detect the alerting tone (if active).
    • 1-20 – the Alerting message will be automatically sent after a predefined number of seconds after the call is dialled into wireless network.
  • Minute parameter – define whether the GSM gateway shall consider call duration or count while limiting outgoing calls.
  • Day of deleting statistics in group (every month) – set a day on which statistic data on disconnected calls shall be deleted.
  • Generate virtual ring tone – enable/disable generation of the virtual ringing tone into the VoIP interface.
  • Call length counting – define whether call minutes or seconds shall be counted.
  • After call relax delay – set the time between the end of the current call and start of the next call via one and the same GSM/UMTS module. The recommended value for heavy–traffic installations is 2 seconds!
    • Add random time – use this auxiliary parameter to add random time in seconds. Thus, the resultant time is the sum of the two above mentioned parameters.

Network settings

  • BTS lock – identify the BTS to which the GSM modules shall log in. Restart the selected GSM modules to execute the changes. 

Caution

  • The BTS lock service work with specific GSM modules only (Q55, Q24, GE910, HE910)!
  • If you set a wrong BTS lock, the selected GSM module(s) will not log in to GSM.
  • Network operator code (MCC+MNC) – set the local mobile network provider code manually. If you do not enter a value, the provider will be selected automatically.
  • Number of registration attempts – set the count of network registration attempts if the network rejects to register the +CREG:3 SIM responses. 
  • Delay after registration denied (1-600) [s] – set the registration delay, i.e. the timeout within which the network may send another response after rejecting +CREG:3.
  • Timeout for registration (10-600) [s] – set the maximum timeout for the module to wait for network login in the NWAIT mode.
  • Next try for registration (0=off, 1-720) [m] – set the interval between relogging attempts.
  • Enable USIM and SIM Application Toolkit – enable/disable the USIM and SIM Application Toolkit services for the Telit HE/GE 910 modules.

Disconnect Call

Set the rules for automatic disconnection of outgoing calls to a wireless network.

  • SIM limit exceeded – automatic call disconnection when the active SIM card call limit is exceeded.
  • Time limit exceeded – automatic call disconnection when the active SIM card time of use is exhausted. 
  • No ALERTING before CONNECT – automatic call disconnection when the gateway receives the call connect message without alerting.

Send CLIP from VoIP to GSM/UMTS

  • Transfer CLIP to GSM/UMTS – enable/disable the function.
  • Separating char – define the CDN/CLIP separating character.
  • Modify (‘–’ remove one digit) – change the CLIP. The ‘–’ character is used for deleting one character from the left. 

Caution

  • The Send CLIP from VoIP to GSM service must be supported by the GSM/UMTS provider's network. Otherwise, the call may be rejected by GSM/UMTS!

GPRS activation

  • APN string – define the Access Point Name (APN) for GPRS connectivity.

Basic settings

  • Roaming enabled for network code – international code for the roaming enabled network. The code consists of the following two numbers:
    • MCC – Mobile Country Code – national code (Czech Republic – 230, e.g.)
    • MNC – Mobile Network Code – GSM network code (T–Mobile 01, O202, Vodafone 03, e.g.)

Hence, the T–Mobile CZ international code is 23001. Leave the field empty to disable roaming.

String

Note

<empty>

Roaming disabled

2300

Roaming disabled (five digits at least)

23002

Roaming enabled for the 23002 (MCC+MNC) network

230XX

Roaming enabled for the 23000 – 23099 (MCC+MNC) network

XX001

Roaming enabled for the 00001 – 99001 (MCC+MNC) network

XXXXX

Roaming enabled for any network

Note

  • Before enabling roaming, please use your mobile phone to make sure that the GSM/UMTS searching priorities have been set properly on the SIM card.

Caution

  • Calls via a roaming network may increase your telephone call costs!
  • CLIR – enable/disable presentation of the SIM CLI on the called party's telephone. CLIR is recommended for the SIM card inserted in the GSM module to avoid CallBack problems.

Caution

  • The Temporary CLIP enable and Temporary CLIR enable services must be supported by the GSM/UMTS provider's network. Otherwise, the call may be rejected by GSM/UMTS!
  • Maximum number of called minutes – define the maximum count of minutes to be called within a month via the selected SIM card.
  • SMS messages number – define the maximum count of SMS messages to be sent within a month via the selected SIM card.
  • Day of restoring call limit and delete statistics – select a day in a month on which the Max count of called minutes and Count of SMS messages statistics shall be deleted.
  • First count – set the length of the first pulse after which the pulse counting change starts as defined in the Next count parameter.
  • Next count – set the length of one pulse in seconds after the time defined in the First count parameter elapses.

Note

  • Set the two parameters above (First countNext count) properly to count free minutes on SIM cards correctly. These parameters are used for limiting outgoing calls depending on free minutes. The CDRs contain real data.
  • Day limit of called minutes – set the maximum count of minutes to be called within a day via the selected SIM card.
  • AOC sending interval – set the interval to send AOC messages

Time limits

There are two SIM use time limits in a GSM group.

 

Call tariffs

Use this function to assign up to four independent free minute counters to a group of GSM modules (SIM cards), e.g.:

Tariff 1 = free minutes for calls to own GSM/UMTS network;

Tariff 2 = free minutes for calls to other GSM/UMTS networks;

Tariff 3 = free minutes for calls to fixed network;

Tariff 4 = free minutes for calls within a closed user group (VPN).

Complete the LCR table (assign prefixes to tariffs) properly to make full use of this function. If routing to a tariff is not used, the global free minutes function will be used in the LCR table.

  • Free minutes - set the free minutes for the tariff offered by the provider.
  • Transferred minutes - set the maximum count of free minutes to be transferred to the next period.
  • AOC sending interval – set the interval to send AOC messages.
  • Day of restoring free minutes - set a day on which the free minute counters will be reset automatically. Select every 24 hours, a day in a month, or a day in a week.
  • Week of restoring free minutes in month - set a week in which the free minute counters will be reset automatically. Set the week number only if the restoration takes place every other Friday in a month, e.g., or keep Every for the other cases.

Caution

  • The recommended free minute counter value is X–5, where X is the number of free minutes obtained from the GSM/UMTS provider. Thus, you can avoid exceeding limits.
  • The manufacturer is not responsible for additional call costs incurred as a result of exceeding your GSM/UMTS provider's free minute/SMS limit.

GSM Incoming Groups

2N® BRI gateway allows you to work with two groups of incoming calls. You can set different values for each of them.

 

General settings

  • Mode – set how the gateway shall process incoming calls from the GSM network.
    • Reject incoming calls – all incoming calls from the GSM network are rejected automatically.
    • Ignore incoming calls – all incoming calls from the GSM network are ignored. The calling party hears the check ring tone.
    • Accept incoming calls + voice message – incoming GSM calls are accepted by the gateway and, if programmed so, DTMF with a voice message is activated for them.
    • Accept incoming calls + dialtone – incoming GSM calls are accepted by the gateway and, if programmed so, DTMF with a simulated second dialtone is activated for them.
    • CallBack after ring / Reject – CallBack will be made if the CLIP is included in the CallBack table. The other incoming calls will be rejected.
    • CallBack after ring / Ignore – CallBack will be made if the CLIP is included in the CallBack table. The other incoming calls will be ignored.
    • Report to PC + voice message – the GSM gateway sends information on the incoming call to a PC equipped with call routing application. If programmed so, DTMF with a voice message is activated for the incoming call.
    • Report to PC + dialtone – the GSM gateway sends information on the incoming call to a PC equipped with call routing application. If programmed so, DTMF with a simulated second dialtone is activated for the incoming call.
  • Minimum digits in DTMF – set the minimum count of digits to be requested by the gateway for DTMF.
  • Maximum digits in DTMF – set the maximum count of digits to be accepted by the gateway for DTMF.
  • Timeout for entering DTMF digits [s] – set the timeout for which the GSM gateway shall wait for the first/next DTMF digit. If you select '0', the incoming call will be automatically connected to the numbers included in the List of called numbers.
  • Day of deleting GSM group statistics – set a day in a month on which the incoming call statistics shall be deleted.
  • Prefix before DISA – set a numerical prefix to precede DTMF.
  • CLIP – use this parameter to modify the incoming CLIP from GSM/UMTS. For international codes, '+' will be removed automatically. Use '–' to remove a digit. Examples (CLIP in GSM: +420600123456):

Parameter

CLIP to VoIP/PRI 1

Note

Null

420261301500

No CLIP change

+

+420261301500

Add + before CLIP

00

00420261301500

Add 00 before CLIP

0261301500

First two digits removed from CLIP

–––99

99261301500

First three digits removed from CLIP, prefix 99 added

  • Looping of voice message – set the voice message playing time.

Send CLIP from GSM/UMTS to VoIP

  • Transfer CLIP from GSM/UMTS – enable/disable the function.
  • Separating char – set the separator for the SIM card CLIP and ID of the extension to be called.
  • Modify – modify the extension ID.

Caution

  • The Send CLIP from VoIP to GSM service must be supported by the GSM/UMTS provider's network. Otherwise, the call may be rejected by GSM/UMTS.

Others

  • Time to keep CLIP in table – set the record keeping time for AutoCLIP routing.
  • Add record only for unconnected call – enable storing of unconnected outgoing calls in the AutoCLIP table only.
  • Delete record for connected answer – enable deletion of an AutoCLIP record in the case of successful CallBack.
  • Skip DTMF for numbers not in CLIP Routing table – enable this option to set DTMF to the incoming calls only whose CLIP is included in the CLIP Routing table. Make sure that the called number table includes one record at least to make the function work. 
  • Skip list of called numbers after failed call to wanted number – enable this function to disable forwarding of incoming calls to the numbers included in the List of called numbers if rejected after DTMF.
  • Auto end to CallBack request – enable that the incoming call whose CLIP meets the CallBack settings will be rejected. If not, the call will be ignored. The CallBack function will be retrieved after call end in both the cases.

List of called numbers

List of numbers to be dialled if DTMF dial–in was not made. Search the table from top to bottom. If the called user is inaccessible, use the following table record.

Prefixes

Use this window to adapt the gateway to calling to various GSM providers' networks. Set the call routing rules based on prefixes for up to sixteen groups.

 

Prefix list 1–16

Sixteen prefix groups to be assigned in the LCR table. 

 

Basic settings

  • GSM network ID – set the prefix list user code for easier orientation in the LCR.
  • Default count of digits – default length of the number to be dialled into the GSM/UMTS networks for routing via the selected prefix list. Use this parameter in case the Digits count is not included in the Accepted prefixes table.

Note

  • The number to be dialled to the GSM/UMTS network must meet the Count of digits condition.
  • For VoIP calls, the count of digits to be dialled must be equal to or higher than the value set in the Count of digits.
  • For GSM/UMTS calls by overlap dialling via the BRI NT/TE interface, the Count of digits defines the maximum count of the digits to be dialled.
  • For GSM/UMTS calls by block dialling via the BRI NT/TE interface, the Count of digits is ignored.

Table of replaced prefixes

Use this table to replace the prefix of the received number ('00' with '+', e.g.). You can only add or remove the prefix. This change is made before the prefix is searched for in the prefix table. Be sure to keep the „/" record in the table for a proper function.

Note

  • The maximum count of records in the Table of replaced prefixes is 14 for each prefixlist.

  • The maximum table record size is 9 characters for the prefix and 9 characters for the replaced number.

Table of accepted prefixes

List of prefixes of called destinations to which the selected prefix list applies.

Note

  • The maximum count of records in the Table of accepted prefixes is 138 for each prefixlist.

  • The maximum table record size is 9 characters.

LCR Table

Table of outgoing Least Cost Routing (LCR) rules. Every outgoing call from the source interface is routed to the destination interface according to this table. For a call, the gateway checks the lines and if the called number prefix matches the prefix in the selected network list and the current time value is within time limitation limits, the call will be routed via the defined GSM group(s) or BRI 1/BRI 2/VoIP interface.

  • From (channels/groups) – source channels or groups via which calls are routed to 2N® BRI gateway.
    • GSM ALL – any of the GSM incoming groups can be used for call routing.
    • GSM GRP1-2 – define one GSM incoming group or a range of GSM incoming groups via which call routing will be enabled.
    • BRI1 (pxx,P) – any of the BRI1 channels can be used for call routing.
    • BRI1 (p1-2) – define one BRI1 channel or a range of BRI1 channels via which call routing will be enabled.
    • BRI2 (exx,E) – any of the BRI2 channels can be used for call routing.
    • BRI2 (e1-2) – define one BRI2 channel or a range of BRI2 channels via which call routing will be enabled.
    • VoIP (vxx,V) – calls are routed via the VoIP interface.
  • Prefix list – prefixes to be used for a selected LCR row. Set up to 16 prefix lists.
    • Prefixlist 1-16 – call routing will obey the rules set in Prefix list 1-16. The count of digits to be dialled is governed by the Prefixes setting. 
    • All prefixes – all the prefix lists are permitted. Prefix lists 1-16 will be searched from 1 to 16. The first match will be used routing. For incoming GSM calls, the Count of digits from the GSM incoming group parameter will be applied.
    • Number of digits – all the prefixes are permitted. Routing is only limited by the count of digits to be dialled.
  • Time limitation – time validity limitation for a selected LCR row.
  • Weekend usage – enable/disable a row on weekends.
  • Call duration limit – set the maximum duration (minutes) for an outgoing call to the GSM/UMTS network.
  • Groups – define the outgoing GSM groups or interface via which outgoing calls will be routed from 2N® BRI gateway. If the defined interface is inactive or the outgoing GSM group tariff is exhausted, the next row will be applied.
    • GSM groups 1-2 – outgoing GSM groups. Set the tariff to be used. Refer to the GSM outgoing groups subsection for details.
    • BRI1 (P) – ISDN BRI1 interface. Refer to the ISDN parameters subsection for details.
    • BRI2 (E) – ISDN BRI2 interface. Refer to the ISDN parameters subsection for details.
    • VoIP (V) – VoIP interface. Refer to the VoIP parameters subsection for details.
  • Tariffs – select the tariff group (free minute counter) to be used for the outgoing call. Refer to the GSM outgoing groups subsection for details.
  • Ignore tone detection in last group – having detected a user defined tone (refer to GSM basic settings), the GSM gateway automatically terminates the call and seeks for another call setup way. If this parameter is activated, the GSM gateway ignores the tone detection results and sets up a call when this is the only possible call establishing way.

Note

  • If you use tariff routing, set the tariffs properly in the GSM outgoing groups subsection.
  • The maximum count of LCR table records is 64.
  • The rules are applied to calls sequentially, starting from the first rule. If all the set rules are met in a row, the call is routed according to the row.
  • Call routing from BRI1/BRI1, BRI2/BRI2 and VoIP/VoIP is not supported. Such calls will be rejected by 2N® BRI gateway.

CLIP Routing Table + CallBack

Use the table to set a fixed CLIP assignment of incoming CLIPs from GSM to the numbers of extensions to which incoming calls are routed automatically. Also, set the CLIP list in the GSM network for which CallBack is enabled.

  • GSM number (CLIP) – user Id in GSM/UMTS.
  • Used service
    • Autodial – enable/disable CLIP routing for the selected CLIP.
    • Reject call – reject call for the selected CLIP.
    • Ignore call – ignore call for the selected CLIP.
    • Tone dial-in – incoming GSM calls is accepted + dialtone is activated for the selected CLIP.   
    • DISA message dial-in – incoming GSM calls is accepted + voice message is activated for the selected CLIP. 
  • Dial to VoIP/ISDN – VoIP destination number for the CLIP routing function.
  • Auto CallBack – enable/disable the CallBack function for the selected CLIP according to the Gateway configuration / GSM basic settings / Calls / Source interface for CallBack setting.
  • Call duration limit  – set the maximum call duration (minutes).

Tip

  • CallBack detects the CLIP from right to left. Thus, configure 10 rules for all the incoming CLIPs to make CallBack work properly. Each of the rules must contain one of the CLIPs: 0,1,2,3,4,5,6,7,8,9.

Note

  • Remember to activate the CallBack mode in the GSM incoming groups for a proper function.
  • The maximum count of CLIP routing table records is 96.

Mobility Extension

Use this table to register the Mobility Extension users.

  • Name – user name for calling to VoIP.
  • User – registration user name for VoIP.
  • Password – registration user password for VoIP.
  • GSM number (CLIP) – user SIM card CLIP.
  • Follow me function – enable/disable call forwarding to the user mobile station (according to CLIP).
  • SMS at no answer function – enable/disable sending of SMS on missed calls.

Ethernet Configuration

Use this window to set the gateway Ethernet interface.

  • Use DHCP – enable/disable the DHCP client function for 2N® BRI gateway.
  • IP address – fixed IP address (v4) for the 2N® BRI gateway Ethernet interface.
  • Subnet mask – network mask for the 2N® BRI gateway Ethernet interface.
  • Default gateway – IP address (v4) of the IP gateway in the Ethernet.
  • DNS server1 – primary DNS server.
  • DNS server2 – secondary DNS server. Used as a back-up when DNS server 1 is not functional.

Caution

  • Saving wrong values, e.g. DHCP enable, may result in making the 2N® BRI gateway configuration part inaccessible. In that case, reset the GSM gateway to factory values; refer to Subs. 4.1.

Tip

  • If the gateway is in the DHCP client mode, the current values obtained from the DHCP server are displayed in the IP address, Subnet mask and Default gateway items.

Login Configuration

Use this window to set the access password and name for the 2N® BRI gateway web interface. Use the same access data for Telnet connection too.

Caution

  • Change the user name and password during your first gateway configuration to avoid unauthorised access to your gateway configuration!

Note

  •  The username and password may have up to 30 characters in total.

Web Configuration

Set additional parameters for web access to the GSM gateway.

  • Auto logout – set the count of minutes in which the user will be logged out automatically.
  • Enable web session lock – one Admin user may be connected to the GSM gateway at one time. If another duly authorised user tries to log in, the preceding session will be terminated automatically. If you activate this function, no automatic logout will occur and any other access attempts will be blocked.
  • Simple login form – activate this item to change the graphic appearance of the login window into an anonymous look. This function is recommended for direct connection of the GSM gateway to the Internet.
  • Use SMS user for SMS operations on the web – enable/disable a user authorised to send/receive SMS messages only.
  • SMS user name/password – connect a user with the right to receive/send SMS only.

Report Configuration

Use the window to set details for automatic tracing generated by the GSM gateway.

Time Synchronisation

Set the NTP server time synchronisation.

 

General

  • Type - enable/disable synchronisation.
  • Ntp server - NTP server address and port.
  • Timezone - set the time change from UTC.

Summer/Winter Time

  • Automatically switch to summer/winter time - enable/disable automatic winter/summer time transition.
  • Switch to summer time - set the summer time transition date and time.
  • Switch to winter time - set the winter time transition date and time.